Local Network Link Operation for a Traditional Digital Key/Hybrid Telephone System
A traditional Digital Key/Hybrid Office Telecommunications System consists of two (2) major components: 1) the Digital Key Telephone instrument; and 2) the Common Equipment Unit (i.e., the back room or wiring closet equipment) which interconnects the Digital Key Telephones and the external Central Office (C. O.) lines.
The typical office internal telecommunications network uses a "Star Wiring Topology", consisting of "home run wiring", where each individual telephone is connected back to the Common Equipment Unit (CEU) on a dedicated Unshielded Twisted Pair (UTP) cable.
There is an important distinction to be made here between an industry standard analog 2500 type telephone (i.e., Touch Tone.RTM. Telephone) connected to a PBX (the type of CEU) and an electronic Digital Key Telephone connected to a PBX. Like the electronic Digital Key telephone, the analog 2500 type telephone is connected to the PBX by "home run wiring", forming a "Star Wiring Topology", where each individual telephone is connected back to the PBX on a dedicated UTP cable. However, the analog 2500 type telephone uses "in-band" audio channel signaling to communicate to the PBX.
Analog PBX Signaling Methods:
The analog 2500 type telephone is connected to the PBX over the Unshielded Twisted Pair (UTP) cable using an industry standard "Loop Interface". The telephone loop interface port (station port) on the PBX provides a source for "DC Loop Current" and an analog signal channel bandwidth from 300 Hz to 3,400 Hz for audio signal transmission. The standard loop interface provides for two types of signaling to the Common Equipment Unit (CEU) over the UTP cable: 1) Hook Switch State and 2) In-Band DTMF (Dual Tone Multi Frequency) Signals.
When the analog 2500 type telephone is "On Hook", it is in the idle state and no DC loop current is flowing between the associated PBX Station Port and the telephone. When the handset of the analog 2500 type telephone is lifted from its cradle (i.e., goes "Off Hook"), the "Hook Switch" contact is closed and DC loop current flows between the PBX Station Port and the telephone. The loop interface circuitry at the PBX station port monitors the status of the DC loop current (i.e., no loop current flowing; or loop current flowing within an acceptable range) to determine the state of the analog 2500 type telephone connected to the PBX station port by the UTP cable. No loop current flowing indicates that the telephone is in the "Idle On Hook State" and requires no servicing. The detection of DC loop current flowing, within an acceptable range, indicates that the telephone has gone "Off Hook" and requires servicing.
Through the "On Hook" and "Off Hook" states produced by the analog 2500 type telephone, and the detection thereof by the associated PBX station port, the telephone can communicate (i.e., signal) to the PBX that it requires service. Now that the telephone has signaled to the PBX that it needs to be serviced, it needs a means to communicate to the PBX what type of service it requires. The type of service request is communicated using "in-band" DTMF Signaling. As previously described, the loop interface provides a 300 Hz to 3,400 Hz bandwidth audio channel between the analog 2500 type telephone and the associated PBX station port. The telephone contains a DTMF signal generator and the PBX station port has access to a DTMF signal detector. The DTMF signaling scheme comprises a base of sixteen (16) unique digits, or characters. The composite spectrum of the DTMF signals fall within the 300 Hz to 3,400 Hz bandwidth audio channel allowing the DTMF digits to be transmitted over the loop interface for communicating service requests and address signaling to the PBX. Once the DTMF signal transmissions have subsided, the audio channel bandwidth is available for the transmission of voice signals. Hence the term "in-band" signaling, where the same channel bandwidth is used to transport both the DTMF signaling information and the voice signal information.
Digital Key Telephone PBX Signaling and Switching Methods
The commercially available systems today use vendor proprietary communications links to transport the digitized voice and telephone control signaling between the proprietary Digital Key Telephone and Common Equipment Unit (CEU) over the Unshielded Twisted Pair (UTP) cable. Typically, equipment vendors transport two (2) full-duplex 64 Kbps Bearer Channels and one (1) full-duplex 16 Kbps Signaling D Channel (2B+D) over the communications link between the telephone and the CEU. The two 64 Kbps B Channels are used to support circuit switched digitized voice, or circuit switched data transport, channels. The 16 Kbps D Channel is used to transport telephone control signaling packets and low speed data (e.g., ASCII character transmission from the CEU to the telephone LCD display).
The two (2) 64 Kbps B Channels are capable of transporting digitized voice in the form of 8 Bit PCM (Pulse Code Modulation) words, or other 8 bit digital data synchronously formatted to these Time Domain Multiplexed (TDM) channels. In both cases, the transport of information in a B Channel is on circuit switched bases. The nature of the circuit switched connection is that it is set up when there is information to transport. It provides a constant bandwidth (in this case 64 Kbps per B Channel) and this constant bandwidth is available for the duration of the connection. Finally, the connection is torn down when it is no longer required. This actually describes the typical telephone call. A telephone number is dialed, the connection is made, and a conversation is held for some period of time. The connection is torn down when the conversation has been completed by the user going on hook. Therefore, the B Channels of the Digital Key Telephone are only active when there is a voice or data call in progress. The B Channels are inactive when the telephone is in the idle state.
The electronic Digital Key Telephone uses out-of-band binary signaling bits via the D Channel to exchange signaling packets with the CEU. The signaling packets are used to transport Lamp Status (Key LED States; On, Off, Flash Rate, etc.) and telephone control commands (CODEC Power UP, Speaker On, Enable Speaker Phone Mode, etc.) from the CEU to the telephone. The D Channel signaling packets sent from the telephone to the CEU are used to transport the Telephone Type Identifier, Hook Switch Status and Key Closure information. Unlike the circuit switched connections supported by the B Channels, the D Channel is always active.
When the telephone is idle, the CEU must still have the ability to send status information to the telephone. For example, the CEU must send Lamp Status commands to the telephone in order for the telephone electronics to update the state of the LEDs under the line keys on the telephone. This is necessary because the idle Multi-Line Digital Key telephone must display the status of the incoming lines (Idle, Busy, Ringing, Hold, etc.) by appropriately illuminating the LED under the associated line key. In addition, the CEU needs a means to communicate to an idle telephone that it has an incoming call, i.e., to transmit the commands to turn on the telephone speaker and produce a ringing sound. Likewise, an idle telephone must have a means to communicate to the CEU that it requires servicing, i.e., that it has gone off hook or has selected an outside line on which to make a call.
Telecom/Data Network Integration
The integration of audio, video and computer data for transmission over a single network has been proposed in the past by a number of authors. Proposals have been advanced for transmitting and receiving packetized voice and data with predesignated time slots within each frame and which share the channel capacity, but giving some form of priority to the delay sensitive voice packets. Proposals have also been advanced for accommodating both isochronous (e.g., video) and non-isochronous (e.g., data) over an isochronous network by replacing standard packet (i.e., 10 Base-T Ethernet) transmission techniques with synchronous Time Domain Multiplexed (TDM) transmission scheme. This requires proprietary and complex interface circuitry to be inserted between the standard Ethernet or Token Ring Media Access Controller (MAC) and the physical transmission media. However, the prior art has not fully addressed the integration of telecom and data in the context of the requirements of a small to mid-size office, typically having personal computers, workstations, servers, printers, etc., connected through Unshielded Twisted Pair (UTP) cable in a LAN using standards-compliant packets such as Ethernet, and also having a Digital Key/Hybrid Telephone System with telephone handsets connected to the Common Equipment Unit (CEU) by a separate UTP cable system. Some of the issues and problems in integrating the two networks are addressed below.
The typical office internal telecommunications network uses a "Star Wiring Topology". A 10 Base-T or 100 Base-TX Ethernet LAN deployed in the small to midsize office uses a similar "Star Wiring Topology". Each individual Personal Computer (PC), Workstation or other Ethernet-supported device, is connected to an Ethernet Hub/Switch using a dedicated UTP cable, i.e., "home run wiring". However, the required quality of the UTP cable is a function of the network transmission speed used. 10Base-T Ethernet, which provides a transmission speed of 10 MBps, uses Category 3 cable or higher; 100Base-TX Ethernet provides 100 MBps on Category 5 cable as well as other physical media such as fiber.
Integrating the Digital Key/Hybrid Telephone system with an Ethernet LAN is enabled under the present invention by using Ethernet packets to transport the B Channel and D Channel information to a common device performing the function of a CEU. This packet transmission method is not an issue for the telephone control signaling packets traditionally transported over the D Channel. However, the B Channels provide transmission of Constant Bit Rate (CBR), circuit switched information. Therefore, transforming the B Channels into standard Ethernet packet transmissions requires some form of CBR, circuit switched, channel emulation, which is one feature of the present invention.
The other major difference to be contended with in providing for the 2B+D transmissions of the Digital Key Telephone operating over the Ethernet LAN is the dedicated vs. shared communications link between stations and the common equipment. It was previously noted that the traditional office telecommunications network and the 10Base-T or 100Base-TX Ethernet LAN both use a physical "Star Wiring Topology" to connect stations to the common equipment (i.e., Telecommunications Switch or Ethernet Hub/Switch respectively). However, most telecommunications networks use this network topology to form dedicated point-to-point transmissions between the common equipment and a single station instrument. Ethernet, on the other hand, allows for the transmission of information from multiple station devices attached to a single Ethernet segment.
In the case of 10Base-T and 100Base-TX Ethernet, each station device is connected back to a Hub or Switch by a dedicated UTP cable. Ethernet Hubs are repeater devices, duplicating the signal transmissions received on one station cable to all other station cables connected to the Hub, producing a single shared Ethernet segment for all station devices. The result is the generation and flow of traffic from multiple sources on the same communications link. Ethernet Switches also act as repeating devices, but are selective repeating devices. An Ethernet Switch reads the destination address from the packet header being received on an ingress port and directs the packet only to the associated egress port (or ports, in the case of multicast). The other ports on the switch will not have the packet information transmitted to them, providing an isolated Ethernet segment for each port on the switch. However, the networking topology does allow for a Hub to be connected to a port on a Switch in order to expand the number of network users. Again, the result is the generation and flow of traffic from multiple sources (i.e., all of the stations connected to the Hub) entering a single port on the switch. The expansion capability of the networking technology requires an integrated Digital Key/Hybrid Telephone system and Ethernet IAN to support multiple Digital Key Telephone terminals attached to a single Ethernet LAN segment. This places on an integrated voice/data common device performing the same function as a CEU the additional task of identifying the individual traffic flow types entering a single system port and directing the individual flows to their appropriate destination.
Digital Key Telephone Signaling Requirement
The operation of the traditional Digital Key/Hybrid telephone system depends upon the control signaling transmissions between the telephone and the Common Equipment Unit (CEU). These signaling transmissions provide the communications link between the Call Processing/Feature software executing on the system CPU and the requests made by the user through the Dial Pad and Feature Keys on the telephone. An independent communications link of this type is required between each Digital Key Telephone and the system CPU in the CEU.
These independent communications links in the prior art separate telephone network are supported over the individual dedicated point-to-point cable connections between each Digital Key Telephone and CEU station port interface. It is important to note here that only one (1) signaling channel flows over any individual station cable. Therefore, each physical station port in the system has a dedicated signaling channel. This provides for a relationship between the physical station port and the station signaling channel for the telephone connected to that port, providing a means for the system software to uniquely identify the associated telephone.
A dedicated signaling channel to each telephone is required to provide a communications link between the Call Processing/Feature software and the requests made by the user through the Keys on the telephone. In the case of multiple Digital Key Telephones connected back to the common equipment to be described over an Ethernet segment, there is not the direct association of the physical system port for defining a signaling channel dedicated to a telephone. Therefore, the establishment of more sophisticated logical signaling channel links to multiple telephones over an Ethernet segment is enabled under the present invention for the exchange of signaling information between the individual telephones and the system CPU in the common device performing the function of the CEU. The method and apparatus for establishing such links is another feature of the present invention.
Quality of Service (QoS) Requirement for Delay Sensitive Data
The most significant element in providing for the transmission of data, voice and videoconferencing over a single network fabric is that of the transport control techniques required to provide the guaranteed Quality of Service (QoS) for audio, video and other delay sensitive data. Depending on the application, bandwidth in and of itself may not be the dominant issue. For example, why should there be any concern about bandwidth when transporting a 64 Kbps digitized voice (PCM) channel over a lOMBps Ethernet segment? Surely there is enough available bandwidth to transport the 64 Kbps PCM information over the segment. Unfortunately, contention between real-time audio and/or video applications and computer file transfer applications for access to the LAN segment causes a problem with real-time transmissions. This contention causes unacceptable latencies to be encountered by packets carrying delay sensitive data, queued waiting to enter the media, while file transfer packets are using the media. This is of particular concern in the case of 10 Mbps Ethernet (10Base-T), where computer file transfers can be using the maximum Ethernet packet size of 1518 bytes. Accounting for the Preamble, Start of Frame Delimiter (SFD) and the Inter-Packet Gap (IPG), a single maximum size packet will occupy the media for 1.23 ms. The latency caused by the transmission of these maximum size packets rapidly consumes the Round-Trip Echo Path Delay specification of 2.0 ms for Digital to Digital Connections in a Digital Key/Hybrid telephone system.
Additional latency can be introduced to packet transmissions by the Ethernet media access control characteristics for the Carrier Sense Multiple Access with Collision Detection (CSMA/CD) access method. Packet collisions on the media caused by the asynchronous transmissions from multiple station devices attached to the media require the retransmission of corrupted packets. When a collision is detected, the transmitting stations back off, select a random delay, execute the delay and transmit again. This process of detecting collisions and re-transmitting packets increases the latency for all packets traversing the network. Packet to collisions and the resulting increased latency become a significant problem in poorly designed or over-subscribed networks (i.e., networks improperly deployed or networks with too many users per segment).
Our proposed Switched Ethernet implementation of an integrated voice/data system reduces the latencies caused by packet collisions on the media and assists in developing a QoS transport technique by isolating collision domains. Switched Ethernet improves network productivity by segmenting network traffic and providing private 10 Mbps (10Base-T) or 100 Mbps (100Base-TX) access to the desktop. However, the requirement for a truly integrated communications system is to provide for all communications needs over a single network fabric to the desktop. The single connection to the desktop dictates that, as a minimum, a Digital Key Telephone and the user's computer or Workstation must share the same LAN segment to the desktop. Therefore, working in a Switched Ethernet environment may greatly improve, but does not eliminate, the problem of having multiple station devices generating independent, and in this case incompatible, traffic streams over the same LAN segment.
Packet queuing delays within the Ethernet Switch also add latency to packet transmissions producing an additional impairment to providing a guaranteed QoS for delay sensitive data. Traditional switch designs have used First-In-First-Out (FIFO) queuing to order the flow of traffic through the switch. Packets leaving a port are organized in the order in which they were received. No special treatment is given to packets from traffic flows that are of higher priority or are more delay sensitive. If a number of packets from different traffic flows are ready to forward, they are handled strictly in FIFO order. When a number of smaller packets are queued behind a longer packet, then FIFO queuing results in a larger average delay per packet than if the shorter packets were transmitted before the longer packets. Guaranteed QoS is not something that is practically supported with the FIFO queuing model.
A number of switch designs have implemented multiple output queues and scheduling algorithms like Weighted Fair Queuing (WFQ), to determine when a packet needs to be serviced in order to improve individual traffic flows. However, traffic from different flows interfere with one another and just adding a priority FIFO queue does not isolate the behavior of each traffic flow. When congestion occurs, the scheduling algorithm must distribute multiple priority traffic flows through the priority FIFO queue, again introducing the latencies associated with the traditional FIFO queuing model. If on the other hand, the switching mechanism provides for prioritizing traffic flows through dynamically allocated flow queues dedicated to each active traffic flow, serviced by priority scheduling algorithms, the inherent problems with the FIFO queuing model are resolved. This scheme allows for traffic streams to be forwarded from the switch independently of the order in which the packets arrive. When the switch has more bandwidth than traffic requires, all traffic can be serviced equally. However, when congestion occurs, the priority scheduling algorithms ensure that packet streams are forwarded according to their minimum guaranteed QoS parameters. It is important to note that either Layer 2 or Layer 3 protocols can be used to establish and control the Priority Flow Queues. This allows for the development of very versatile and powerful switching algorithms.
Developing an integrated voice/data communications system based on state of the art Ethernet Switching technology provides private 10 Mbps (10Base-T) or 100 Mbps (100Base-TX) access to the desktop with individually regulated traffic flows. Versatile switching and scheduling algorithms can be implemented to provide a guaranteed delay QoS for individual packet streams through the switch. However, the incremental resolution of the traffic flow control is limited to a discrete packet base. Mixed simultaneous traffic flows of large packets carrying computer file transfer information and small packets carrying delay sensitive information, on a limited bandwidth port (e.g., 10 Mbps), still present an impairment to providing a guaranteed QoS for the delay sensitive information.